IP Multimedia (IPMM) services provide a dynamic combination of voice, video, messaging, data, etc, within the same session. By growing the numbers of basic applications and the media which it is possible to combine, the number of services offered to the end users will grow, and the inter-personal communication experience will be enriched. This will lead to a new generation of personalised, rich multimedia communication services, including so-called “combinational IP Multimedia” services.
IP Multimedia Subsystem (IMS) is the technology defined by the Third Generation Partnership Project (3GPP) to provide IP Multimedia services over mobile communication networks. IMS provides key features to enrich the end-user person-to-person communication experience through the integration and interaction of services. IMS allows new rich person-to-person (client-to-client) as well as person-to-content (client-to-server) communications over an IP-based network. The IMS makes use of the Session Initiation Protocol (SIP) to set up and control calls or sessions between user terminals (or user terminals and application servers). The Session Description Protocol (SDP), carried by SIP signalling, is used to describe and negotiate the media components of the session. Whilst SIP was created as a user-to-user protocol, IMS allows operators and service providers to control user access to services and to charge users accordingly. Other protocols are used for media transmission and control, such as Real-time Transport Protocol and Real-time Transport Control Protocol (RTP/RTCP), Message Session Relay Protocol (MSRP), and Hyper Text Transfer Protocol (HTTP). IMS requires an IP based access network which for example could be a 3GPP Packet Switched (PS) network, or some other access network such a fixed broadband or WiFi network.
A fundamental requirement for real-time service provision is the seamless handover of services for subscribers roaming across cell boundaries of the radio access network (RAN). Traditional circuit switched (CS) based call services have been designed to meet this requirement. In the case of 2G and currently implemented 3G networks, PS real time handover with low latency is not provided for although service continuity is achieved at the terminal side by ordering a session to be moved from one cell to another, i.e. there is no prepare phase to shorten latency when moving cell.
Real time PS handover is standardized in 3GPP for 3G networks, but the feature has not yet been deployed. It is expected that when High-Speed Downlink Packet Access (HSDPA) is deployed, or shortly thereafter, the mechanisms needed for fast PS handover will be also be deployed. In the initial implementation stage, roll-out of this feature across 3G networks will inevitably be patchy. For 2G networks, fast and efficient PS handover procedures in the packet switched (PS) domain within the 2G network (and between 2G and 3G networks) have only recently been standardized in 3GPP TS 43.129 for GSM/EDGE networks but are not yet deployed. Support for PS handover in 2G networks is never likely to be comprehensive (if implemented at all), yet handover of PS calls would be desirable as 2G networks will continue to provide a fallback network for 3G subscribers in the case of limited 3G network coverage. It can also be expected that the next generation radio and core network which are currently being specified under the name LTE (Long Term Evolution) and SAE (System Architecture Evolution) in 3GPP will also have limited coverage, and that these networks will also require fallback to 3G and 2G networks.
It is expected that in the future a major user of PS services will be Voice-over-IP (VoIP) applications. VoIP calls will be particularly sensitive to even relatively minor service interruptions caused by inter-cell handovers. As long as a terminal engaged in a VoIP call can perform PS handover to another cell (the “target cell”), the interruption can be kept short enough to avoid any noticeable drop in perceived quality. However, if either the current cell or the target cell do not support PS handover, a noticeable interruption is likely to occur as packets will be lost during the transition period. Consequently, until all RAN cells support PS handover, the provision of IMS services such as voice and video calls utilising the PS domain are likely to result in users receiving a reduced quality of service when crossing cell boundaries.
Mobile CS services based on GSM and WCDMA radio access are a world-wide success and allow obtaining telecommunication services with a single subscription in almost all countries of the world. Also today, the number of CS subscribers is still growing rapidly, boosted by the role out of mobile CS services in dense population countries such as India and China. This success story is furthermore extended by the evolution of the classical MSC architecture into a softswitch solution which allows using packet transport infrastructure for mobile CS services.
Recently the 3GPP work item “Evolved UTRA and UTRAN” (started in summer 2006) defined a Long-Term Evolution (LTE) concept that assures competitiveness of 3GPP-based access technology. It was preceded by an extensive evaluation phase of possible features and techniques in the RAN workgroups that concluded that the agreed system concepts can meet most of the requirements and no significant issue was identified in terms of feasibility.
LTE will use Orthogonal Frequency Division Multiplexing (OFDM) radio technology in the downlink and Single-Carrier Frequency Division Multiple Access (SC-FDMA) for the uplink, allowing at least 100 Mbps peak data rate for downlink data rate and 50 Mbps for uplink data rate. LTE radio can operate in different frequency bands and is therefore very flexible for deployment in different regions of the world.
In parallel to the RAN standardization 3GPP also drives a System Architecture Evolution (SAE) work item to develop an evolved core network (CN). FIG. 1 illustrates schematically the System Architecture Evolution (SAE) and LTE interfaces. The SAE core network is made up of core nodes, which are further, split into Control Plane (Mobility Management Entity (MME) 21) and User Plane Gateway 22 (Serving Gateway and PDN Gateway) nodes. In the context of the present invention, the terms Access Gateway (AGW) and SAE GW are used to depict both the Serving Gateway and the PDN Gateway nodes and functions. In the terminology currently used AGW contains both User Plane Entity (UPE) and Inter-Access Anchor (IASA) functionality. The MME 21 is connected to the eNodeB 23, 23′ via the S1-MME interface and the AGW 22 is connected to the eNodeB 23, 23′ via the S1-U interface.
Common to both LTE and SAE is that only a Packet Switched (PS) domain will be specified, i.e. all services are to be supported via this domain. GSM (GPRS) and WCDMA however provide both PS and Circuit Switched (CS) access simultaneously.
Hence, if telephony services shall be deployed over LTE radio access and SAE core networks, an IMS based service engine (or similar) is needed. It has been recently investigated how to use LTE/SAE as access technology to the existing Mobile Switching Subsystem (MSS) infrastructure. The investigated solutions are called “CS over LTE/SAE”, or briefly just “CS over LTE” (CSoLTE).
The basic CSoLTE architecture for these solutions is shown in FIG. 2. The Packet Mobile Switching Center (PMSC) 24 can be serving both traditional 2G and 3G RANs and the new CS over LTE based solutions. Packet MSC 24 contains two new logical functions called Packet CS Controller (PCSC) 27 and Interworking Unit (IWU) 28 that are further described in relation to FIG. 3.
Referring now to FIG. 3, the communication between the terminal 31 and the PMSC 24 is based on the standard Gi interface which is also called as a SGi interface in the SAE terminology. This means that all direct communication between the terminal 31 and the PCSC 27 and the IWU 28 in the PMSC 24 is based on Internet Protocol (IP) and that the terminal 31 is visible and reachable using an IP-address via the Access Gateway (AGW) 22. This communication between the terminal 31 and the PMSC 24 is divided into two different interfaces, U8c for the control plane and U8u for the user plane. The U8c is terminated in the PCSC 27 and the PCSC 27 has also an Rx interface to the Policy and Charging Rule Function (PCRF) 33 for authorising of LTE/SAE bearers. The U8u is terminated in the IWU 28.
An example solution for providing CS services over the LTE radio access is called “CS Fallback” and means that the terminal is performing SAE MM procedures towards the MME 21 while camping on LTE access. The MME 21 registers the terminal in the MSC-S 29 for CS based services. When a page for CS services is received in the MSC-S 29 it is forwarded to the terminal 31 via the MME 21 and then the terminal 31 performs fallback to the 2G or 3G RANs 41. Similar behavior applies for Mobile originated CS services and when these are triggered and the terminal 31 is camping on LTE access, it will fallback to 2G or 3G RANs and trigger the initiation of the CS service there.
The CSoLTE control plane protocol architecture between the terminal 31 and the PMSC 24 (i.e. the U8c interface) is shown in FIG. 4. Interposed between the two is the eNodeB 23 and the AGW 22. This architecture is based on IP protocols (IP, TCP, UDP) and an additional tunneling protocol named as U8-Circuit Switched Resources (U8-CSR). This protocol carries the Mobility Management (MM) and all the protocol layers above MM transparently between the terminal 31 and the PMSC 24.
The CSoLTE user plane protocols between the terminal and the PMSC 24 (i.e. the U8u interface) are shown in FIG. 5. EnodeB 23 and AGW 22 are arranged between the two. This architecture is based on IP protocols (IP, UDP, RTP) that are used to transmit the necessary voice and data communicating (e.g. AMR coded voice) between the terminal 31 and the PMSC 24.
No known solutions exist for Handover from traditional CS domain to the CSoLTE based solutions.
It is an object of the present invention to obviate at least some of the above disadvantages and to provide a method and apparatus for triggering the seamless handover of an established connection from a circuit switched domain to circuit switched service over packet switched domain.